[Asterisk-Users] Zone Paging

[John Todd]

>  > I see a lot of chatter in the archives about intercom and paging, but
>>  has anyone addressed zone paging?  Each of the 50 telephones in a large
>>  clinic would be members of one or more paging zones.  Someone could then
>>  page Dr. X in zone #1.  Would this be possible with analog phones?  SIP?
>
>Summarizing from memory only, I believe this has been discussed more then
>once on the list and usually comes down to... what device(s) provide an
>auto-answer extension and includes an audio output jack that can be
>plugged into a paging amplifier?
>
>The two most common suggestions (from memory again) has been:
>  1. Use of the sound card on the asterisk machine (which sort of implies
>     a limit of one paging zone), or,
>  2. Use a sip phone (Cisco 7960 with v6 as one example), configure the
>     phone to support auto-answer, and connect the external headset to
>     the paging amp. (Implies one sip phone per paging zone.)
>
>I've not tried either, so not sure of success/failure rates or problems.
>
>Seems like a fair number of people have problems getting the sound card
>to play nicely with asterisk, and most of the chatter seems to be oriented
>around sound card driver issues, etc.
>
>Don't know if the ata-186 supports auto-answer in current software, but if
>it did, jury-rigging a matching transformer as a source of zone audio
>would not seem like it would be very difficult. (Anyone know whether the
>186 can be configured for auto answer?)
>
>Rich

I have experimented with this in tentative ways in the following 
manner with Cisco 7960 phones with the new 6.1 SIP image which has 
the ability to auto-answer:

- configure members of your group with 7960 phones such that they 
have a line that auto-answers.  If you want to be clever, you can 
make it so that any outbound calls made on that key will lead to your 
paging extension ( use _. )

- create a special "paging" extension, ringable from other lines

- when the paging extension is called, it runs a short AGI script 
that dumps out several .call files that call the members of the 
paging group, and sends them all to a conference call.  The call 
duration is forced to be 20 seconds (AbsoluteTimeout) and all the 
called parties are set to mute.  The caller makes their <20 second 
announcement, and hangs up.  The other phones go silent, and after 
the 20 second timer, they are hung up.

- the caller then has to pause for a few seconds to wait for RTP to 
catch up; this is a minor inconvenience, and users hopefully will 
quickly figure out that they need to pause for a few seconds before 
they start talking.

- you may experience strange out-of-sync jitter if the phones are 
turned up, since each station will be firing up a separate RTP stream 
to the * server.  This is meant for quiet, at-the-desk paging instead 
of the kind of blaring-trumpets-and-cannons paging you get at 
somewhere like a machine shop or an auto body garage

JT